Hi White, It is no problem. This is a small function which is customized by myself for internal calls only in my company. It is not commercial activity. So , it is not something violation.
Best regards, NHSON -----Original Message----- From: asterisk-dev-boun...@lists.digium.com [mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of asterisk-dev-requ...@lists.digium.com Sent: Friday, April 25, 2014 9:02 PM To: asterisk-dev@lists.digium.com Subject: asterisk-dev Digest, Vol 117, Issue 172 Send asterisk-dev mailing list submissions to asterisk-dev@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-dev or, via email, send a message with subject or body 'help' to asterisk-dev-requ...@lists.digium.com You can reach the person managing the list at asterisk-dev-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-dev digest..." Today's Topics: 1. [Code Review] 3479: chan_pjsip: Call pickup test. (Joshua Colp) 2. [Code Review] 3478: chan_pjsip: Add call pickup support. (Joshua Colp) 3. Add new option to Queue function (Nguyen Hoang Son) 4. Re: encoding issues in Asterisk 11.9.0 Now Available (Matthew Jordan) 5. Re: [Code Review] 3478: chan_pjsip: Add call pickup support. (Matt Jordan) ---------------------------------------------------------------------- Message: 1 Date: Fri, 25 Apr 2014 13:05:57 -0000 From: "Joshua Colp" <reviewbo...@asterisk.org> To: "Joshua Colp" <reviewbo...@asterisk.org>, "Joshua Colp" <jc...@digium.com>, "Asterisk Developers" <asterisk-dev@lists.digium.com> Subject: [asterisk-dev] [Code Review] 3479: chan_pjsip: Call pickup test. Message-ID: <20140425130557.5072.51...@sonic.digium.api> Content-Type: text/plain; charset="utf-8" ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3479/ ----------------------------------------------------------- Review request for Asterisk Developers. Repository: testsuite Description ------- This is a modified version of the normal call pickup test which uses chan_pjsip instead of chan_sip to test call pickup functionality. Diffs ----- /asterisk/trunk/tests/channels/pjsip/call_pickup/test-config.yaml PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast2/pjsip.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast2/extensions.con f PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/pjsip.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/features.conf PRE-CREATION /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/extensions.con f PRE-CREATION Diff: https://reviewboard.asterisk.org/r/3479/diff/ Testing ------- I tested the test by running the test. Thanks, Joshua Colp -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/60e43d1 1/attachment-0001.html> ------------------------------ Message: 2 Date: Fri, 25 Apr 2014 13:06:00 -0000 From: "Joshua Colp" <reviewbo...@asterisk.org> To: "Joshua Colp" <reviewbo...@asterisk.org>, "Joshua Colp" <jc...@digium.com>, "Asterisk Developers" <asterisk-dev@lists.digium.com> Subject: [asterisk-dev] [Code Review] 3478: chan_pjsip: Add call pickup support. Message-ID: <20140425130600.5072.92...@sonic.digium.api> Content-Type: text/plain; charset="utf-8" ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3478/ ----------------------------------------------------------- Review request for Asterisk Developers. Repository: Asterisk Description ------- While configuration exists to place PJSIP channels into pickup and call groups the functionality to actually perform a call pickup does not exist. This change adds it. Diffs ----- /branches/12/res/res_pjsip_session.c 413007 /branches/12/channels/chan_pjsip.c 413007 Diff: https://reviewboard.asterisk.org/r/3478/diff/ Testing ------- Ran test and confirmed failed on normal 12. Applied change. Re-ran test and confirmed fixed. Thanks, Joshua Colp -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/e774ec9 3/attachment-0001.html> ------------------------------ Message: 3 Date: Fri, 25 Apr 2014 20:59:41 +0700 From: "Nguyen Hoang Son" <nh...@vasc.com.vn> To: <asterisk-dev@lists.digium.com> Subject: [asterisk-dev] Add new option to Queue function Message-ID: <016c01cf608e$9ac55580$d0500080$@com.vn> Content-Type: text/plain; charset="utf-8" Hi all, I'm using Queue function of Asterisk to arrange calls which is coming to my agents. I want to customize the way asterisk arrange coming call, in other word, is it possible to create a new option instead of using the existing: RINGALL, ROUNDROBIN,... . For example: The incoming call should come to the argent who has the most waiting time (idle time). I find out that the algorithm of each option of Queue is defined in "app_queue.c" in the source code but I don't know how to change, how to add the waiting time as a new option to sort by. This question is quite related to the development of asterisk, so please help if you have any idea or experience on that. Thank you very much. --------------------------- NGUY?N HO?NG S?N M-Commerce Center VASC Software and Media Company - VNPT Addr: No.97 Nguyen Chi Thanh Street, Dong Da District, Hanoi, Vietnam Cell phone: +84 912998101 Skype: hoangsonk49 E-mail: nh...@vasc.com.vn -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/6d6999f 1/attachment-0001.html> ------------------------------ Message: 4 Date: Fri, 25 Apr 2014 09:06:33 -0500 From: Matthew Jordan <mjor...@digium.com> To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Subject: Re: [asterisk-dev] encoding issues in Asterisk 11.9.0 Now Available Message-ID: <CAN2PU+6DF3PF2cqjnSjqhNjLr7kRdsdL5Yp=6eon2fphekw...@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 On Fri, Apr 25, 2014 at 4:32 AM, Walter Doekes <walter+asterisk-...@osso.nl> wrote: > On 23/04/14 18:52, Asterisk Development Team wrote: >> >> --===============4365525224653466459== >> Content-Type: text/plain; charset="us-ascii" >> Content-Transfer-Encoding: 8bit > > ... >> >> * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted >> minus signs (Reported by Jeremy Lain??) > > ... >> >> * ASTERISK-19499 - ConfBridge MOH is not working for transferee >> after attended transfer (Reported by Timo Ter??s) > > ... > > Could you update the `charset` param to "utf-8" the next time? > > Thanks! > Sure - sorry about that! -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org ------------------------------ Message: 5 Date: Fri, 25 Apr 2014 14:14:15 -0000 From: "Matt Jordan" <reviewbo...@asterisk.org> To: "Joshua Colp" <jc...@digium.com>, "Asterisk Developers" <asterisk-dev@lists.digium.com>, "Matt Jordan" <reviewbo...@asterisk.org> Subject: Re: [asterisk-dev] [Code Review] 3478: chan_pjsip: Add call pickup support. Message-ID: <20140425141415.10829.15...@sonic.digium.api> Content-Type: text/plain; charset="utf-8" ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3478/#review11741 ----------------------------------------------------------- Ship it! /branches/12/res/res_pjsip_session.c <https://reviewboard.asterisk.org/r/3478/#comment21534> Blob. - Matt Jordan On April 25, 2014, 8:06 a.m., Joshua Colp wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3478/ > ----------------------------------------------------------- > > (Updated April 25, 2014, 8:06 a.m.) > > > Review request for Asterisk Developers. > > > Repository: Asterisk > > > Description > ------- > > While configuration exists to place PJSIP channels into pickup and call groups the functionality to actually perform a call pickup does not exist. This change adds it. > > > Diffs > ----- > > /branches/12/res/res_pjsip_session.c 413007 > /branches/12/channels/chan_pjsip.c 413007 > > Diff: https://reviewboard.asterisk.org/r/3478/diff/ > > > Testing > ------- > > Ran test and confirmed failed on normal 12. Applied change. Re-ran test and confirmed fixed. > > > Thanks, > > Joshua Colp > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/709ecd8 a/attachment.html> ------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Put in your talk proposal: http://www.bit.ly/speak-astricon2010 asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev End of asterisk-dev Digest, Vol 117, Issue 172 ********************************************** -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev